[..] |
ASTERISK-22000: chan_alsa fails to use input_device=plug:dsnoop |
ASTERISK-22001: Running monitors crash Asterisk when a monitored channel leaves a bridge. |
ASTERISK-22002: Stasis: Split caching from caching topics |
ASTERISK-22003: Crash - signal 6, aborted - assertion failure in pjsip_auth_create_digest from /usr/lib/libpjsip.so |
ASTERISK-22005: Allow a sip peer to accept both AVP and AVPF calls |
ASTERISK-22006: bridges/ast_bridge_playfile: Use a bridge technology callback to resume entertainment sounds. |
ASTERISK-22007: chan_sip: segfault with invalid sdp |
ASTERISK-22008: Config framework docs should display the see-also information in CLI output. |
ASTERISK-22009: Config framework does not handle reloading multiple config files correctly. |
ASTERISK-22011: Crash in confbridge CLI command handle_cli_confbridge_list on NULL channel pointer |
ASTERISK-22014: Crash in libnetsnmpagent |
ASTERISK-22015: 'core restart now' from a secondary remote console resulted in unresponsive CLI and certain commands no longer functioning |
ASTERISK-22017: crash - assertion failure - in pj_sockaddr_get_port when in transport_apply at res_sip/config_transport.c:105 |
ASTERISK-22019: overlap=no in chan_dahdi.conf does not work |
ASTERISK-22020: ooh323 Q931DisplayIE causes Anonymous on phone displays |
ASTERISK-22021: Amaizing CallerID |
ASTERISK-22023: SIP Caller ID - Logic of trust_id_inbound and trust_id_outbound may be off, plus help descriptions may be unclear |
ASTERISK-22025: [patch] Add IPv6 Support To chan_iax2 |
ASTERISK-22026: Peers in realtime not register after "sip reload" using callbackextension |
ASTERISK-22034: Investigate whether or not a Local channel is appropriate during a one-touch parking feature |
ASTERISK-22035: Properly handle media on channels that are swapped into a parking lot's holding bridge |
ASTERISK-22036: Validate that the bridge requested in a Remove Channel from Bridge operation is the bridge the channel is in |
ASTERISK-22037: Fix AMI action AttendedTransfer to use both DTMF Begin and End |
ASTERISK-22038: Create a secondary message router for cached messages |
ASTERISK-22039: Remove the bridged channel pointer from ast_channel |
ASTERISK-22040: Remove the bridge pointer from ast_channel; refactor consumers of the bridge pointer to safely get it through bridge_channel |
ASTERISK-22041: Move the ao2 string container to a more appropriate location |
ASTERISK-22042: Set a cause code on a channel when it is ejected from a bridge |
ASTERISK-22043: Handle DTMF wrap up operations and Hold wrap up operations when a channel is pulled from the bridge |
ASTERISK-22054: module reload of res_sip_outbound_registration.so results in Asterisk crash - assertion failure |
ASTERISK-22056: Asterisk 11.4.0 : Queue RINGNOANSWER wrong ring time when one of the peer becomes unresponsive |
ASTERISK-22060: Assertion triggered in CDR code when called channel is redirected using AMI |
ASTERISK-22061: crash - Dialing chan_gulp/pjsip with explicit SIP URI results in segfault in ast_sip_session_send_request_with_cb at res_sip_session.c |
ASTERISK-22063: Ringback tone is not heard by caller when the call is transferred using blonde transfer in a Local bridge |
ASTERISK-22064: crash - res_sip outbound registration to offline server fails with crash after X attempts - in sip_outbound_registration_response_cb at res_sip_outbound_registration.c |
ASTERISK-22066: Analog phone digit delay when using cannot-complete-as-dialed catch-all in dialplan |
ASTERISK-22067: Properly handle implied Accept types for SIP event packages |
ASTERISK-22068: chan_iax will not accept calls without destination |
ASTERISK-22069: unexpected absence of connectedline update if set before answering |
ASTERISK-22070: Investigate PJMEDIA's implicit codec negotiation behavior |
ASTERISK-22071: chan_sip doesn't respect Via ..completely |
ASTERISK-22072: 'I' Option Not Supported - bridging core lacks support to suppress COLP updates when joining a bridge |
ASTERISK-22073: Asterisk 12 CEL - inconsistent ordering between APP_END and HANGUP |
ASTERISK-22074: AddQueueMember Causes AMI Disconnect |
ASTERISK-22075: Mutliple SIP NOTIFIES at "sip reload" if mailbox is monitored by multiple sip peers |
ASTERISK-22079: Segfault: INTERNAL_OBJ (user_data=0x6374652f) at astobj2.c:120 |
ASTERISK-22081: Type cast causes incorrect schedule interval |
ASTERISK-22082: 180 Ringing not forwarded out on inbound channel during parallel Dial due to Dial waiting on provisional response from outbound channels |
ASTERISK-22083: res_musiconhold segfault in free, in moh_scan_files |
ASTERISK-22084: ARI: Media operations (playback, start/stop moh) initiated on a channel don't work if that channel is in a bridge |
ASTERISK-22087: res_sip WARNING 'Unable to subscribe extension ...' should specify the type of subscription if possible |
ASTERISK-22089: res_sip - Need log message indicating when Asterisk fails to find an AOR to match an inbound registration |
ASTERISK-22091: Can't build res_rtp_asterisk - libuuid not found after applying patches from 11.5.0 |
ASTERISK-22092: Gulp blond transfers result in channels not being hung up properly |
ASTERISK-22093: Deadlock due to locking inversion between PBX context lock and channel lock while reloading dialplan through pbx_lua |
ASTERISK-22094: res_sip - transport config object can't be modified on module reload - let the user know |
ASTERISK-22096: chan_pjsip - when dialing an endpoint that is misconfigured, debug could be more helpful in regards to whether Asterisk found a matching AOR or not |
ASTERISK-22098: Dahdi BRI OverlapDial: Digits in ISDN Call setup message do not show up in ${EXTEN} |
ASTERISK-22101: res_sip_endpoint_identifier_ip needs debug to indicate when it's working |
ASTERISK-22104: Bridge API Enhancements - update the Native RTP bridge to better manage channels |
ASTERISK-22105: [patch] res_pjsip - xml doc change for transport config object - remove warning and add text regarding Asterisk restart |
ASTERISK-22107: Bridge API Enhancements - refactor and redesign ast_bridge_featuresremove interval hooks from ast_bridge_features |
ASTERISK-22108: [patch] res_pjsip - xml doc revision for 'auth' config object and 'auth_type' config option |
ASTERISK-22112: res_sip - 'contact_status' config object, do we need xml config docs for it when it isn't manually configurable? |
ASTERISK-22114: [patch] res_pjsip - 'domain_alias' config object XML help doesn't make it clear that the name used for the object is the domain alias |
ASTERISK-22116: Bridge API Enhancements: Add the ability to selectively remove bridge feature hooks |
ASTERISK-22117: Bridge API Enhancements - add lonely flag support to eject non-participating channels from a bridge |
ASTERISK-22118: [patch] res_pjsip - xml doc revisions for 'aor' config object and a few of its options |
ASTERISK-22120: Missing debug strings |
ASTERISK-22123: Mixmonitor does not create the file and call is muted |
ASTERISK-22125: PRI_EVENT_NOTIFY option to give a "code of event" |
ASTERISK-22126: Bridging: Memory leak for channels that hang up if they were in the bridging system |
ASTERISK-22127: Bridges/chan_sip: Directmedia settings not respected for setting up native_rtp bridge technology |
ASTERISK-22128: ARI/bridges: chan_sip channels with directmedia=yes - Asterisk doesn't retake the media when the technology changes from native rtp |
ASTERISK-22129: Some chan_dahdi protected function renaming. |
ASTERISK-22130: Bridge API Enhancements - refactor Bridging API to hide protected functions and break up large file structure |
ASTERISK-22131: Update the make dependencies script to pull, build, and install the correct pjproject |
ASTERISK-22132: Perform a complete rename of chan_gulp to chan_pjsip |
ASTERISK-22133: Document realtime schemas for chan_pjsip objects |
ASTERISK-22134: Bridge API Enhancements - refactor and destroy as much of features.c as possible |
ASTERISK-22135: res_sip: Restructure ast_sip_endpoint to have better structure |
ASTERISK-22136: API Improvements: rename stasis_http to ARI |
ASTERISK-22138: res_parking: Restore the parking unit tests |
ASTERISK-22139: event.c: Remove as many types as possible and as much dead code as possible |
ASTERISK-22140: BridgeInfo Action: return full cached channel snapshots with events |
ASTERISK-22141: Sounds indexer: improve CLI commands |
ASTERISK-22142: res_parking: fix module unloading |
ASTERISK-22143: res_sip: expose threadpool options as general settings; investigate thread shutdown issues |
ASTERISK-22144: res_sip_dtmf_info: Support sending of 'raw' DTMF |
ASTERISK-22145: res_pjsip: Update the .conf files with real default examples |
ASTERISK-22147: res_sip_private.h - move OPTION request "qualifying" into registrar |
ASTERISK-22148: res/stasis: rename translation units; consider a rename of Stasis App to something else |
ASTERISK-22149: ARI: Update/create automatic library code generation tools in git to use PyStache/Mustache templates |
ASTERISK-22150: Channels/chan_pjsip: sending a CANCEL request during a pending INVITE request can cause a crash |
ASTERISK-22176: Google Voice Incoming And Outgoing Calls Fail With Certain Google Voice Accounts |
ASTERISK-22177: Upgrade.txt from 11.4 to 11.5 does not show requirement for new dependency |
ASTERISK-22178: Enquiry about channel bandpass filter (narrowband and wideband) in codecs implementations G.711 and G.722 |
ASTERISK-22179: Update copyright headers - they're so last year |
ASTERISK-22181: Asterisk REST API - Implement POST /recordings/live/{id}/{control} |
ASTERISK-22183: [patch]Fake acceptance of AOC MESSAGEs |
ASTERISK-22184: SRTP not detected |
ASTERISK-22186: CLONE - [patch] Add kick all capability to app_confbridge's CLI command 'kick' |
ASTERISK-22188: Asterisk crashes inside pjsip when an address is unreachable |
ASTERISK-22189: Wrap up time is ignored for queue members who are members in multiple queues |
ASTERISK-22191: wrong timing for periodic announcement in queue |
ASTERISK-22192: [patch] Allow voicemail forwards with ODBC backend when format differs from attachfmt column |
ASTERISK-22193: Add a to_ami() callback for parking stasis messages |
ASTERISK-22195: Faxdetect sets callerid to be uniqueid |
ASTERISK-22196: Asterisk crashes setting RTP source address to the browser |
ASTERISK-22197: [patch] Queuelog EXITWITHKEY only two of four parameters |
ASTERISK-22200: Some users of ast_cond_wait fail to properly check the predicate condition they guard |
ASTERISK-22201: Despite not bridging early media in a parallel Dial, we forward 183 Session Progress back to the caller |
ASTERISK-22205: incoming DID not wanted |
ASTERISK-22206: No audio on Asterisk 11 when calling from Chrome to PSTN and calls go to voicemail/another call leg |
ASTERISK-22208: Extremely high CPU load and segfaults (in ast_hashtab_start_traversal) on a reload |
ASTERISK-22209: Bridge API Enhancements - Make dial, queue, etc. add their features to the bridge DTMF features datastore instead of override them. |
ASTERISK-22210: 'Bad Magic Number' with 160 Confbridge or more. |
ASTERISK-22211: Crash in native RTP bridge when a channel leaves while switching to core bridge during smart bridge operation |
ASTERISK-22212: Time Out Failures in ACL tests |
ASTERISK-22213: Multiple CDR Test Failures |
ASTERISK-22214: Testsuite test callparking fails due to Python exception on unexpected AMI event |
ASTERISK-22215: All SIP blind transfer tests are failing due to time out |
ASTERISK-22216: Test hang/possible deadlock in tests/channels/SIP/sip_custom_presence/multiple_state_change |
ASTERISK-22217: TestSuite sip_hold test fails in SIPp scenarios on unexpected SIP INVITE requests |
ASTERISK-22218: Crash in chan_gulp/chan_pjsip test incoming_calls_without_auth when attempting to create a channel with a NULL endpoint |
ASTERISK-22219: IAX2 basic_call test fails with CEL/CDR errors |
ASTERISK-22220: Local query option for T.38 faxing test fails |
ASTERISK-22221: The masquerade super-test fails on all Asterisk versions |
ASTERISK-22222: TestSuite: Attended Transfer Feature test fails |
ASTERISK-22236: REGISTER reply send to bad port with nat=yes(or force_rport,comedia) in 11.5.0 |
ASTERISK-22237: [patch] http_shutdown incomplete |
ASTERISK-22238: [patch] astfd and threadstorage debug cli commands are not unregistered |
ASTERISK-22239: [patch] Missing extra line break between peers when running AMI action SIPPeers |
ASTERISK-22240: Asterisk 1.8 func_odbc oracle stored procedure is not working |
ASTERISK-22243: Odd misbehavior (crashes and other such things) in Stasis caching topics |
ASTERISK-22245: WebRTC fails to take calls from Chrome and Mozilla. |
ASTERISK-22246: Asterisk's "T" flag is ignored when used with "r" or "R" flags. (documentation bug) |
ASTERISK-22248: [patch] test_sip_rtpqos corrupts dialogs container |
ASTERISK-22249: [patch] xmldoc.c leaks an attribute |
ASTERISK-22250: after dial 7, i must wait 1 sec before hear second dialtone, i want hear second dialtone at once or at most 0.5 second |
ASTERISK-22251: after dial 7, i must wait 1 sec before hear second dialtone, i want hear second dialtone at once or at most 0.5 second |
ASTERISK-22252: res_musiconhold cleanup - REF_DEBUG reload warnings and ref leaks |
ASTERISK-22253: [patch] pjproject: pkgconfig file generated with DESTDIR |
ASTERISK-22254: [patch] pjproject: include major version number in SONAME |
ASTERISK-22257: [patch] pjproject: make uninstall |
ASTERISK-22258: Queue crashes when publishing message to Stasis after ringing busy Agent |
ASTERISK-22259: [patch] cel segfault on invalid cel.conf |
ASTERISK-22262: Asterisk sends SIP 488 to T38 fallback re-invite |
ASTERISK-22263: [patch] 'queue add member ...' help text update |
ASTERISK-22269: app_meetme: wrong bit value for CONFFLAG_DONT_DENOISE |
ASTERISK-22270: Attended transfer to a Queue stops music on hold on caller |
ASTERISK-22272: [patch] Unexepected behaviour with adaptive odbc filter |
ASTERISK-22273: Asterisk crashes when accessing http://localhost:8088/ari/asterisk/variable |
ASTERISK-22275: [patch] T.38 Passthrough broken if peer doen't report T38MaxBitRate |
ASTERISK-22276: Test test_hashtab_thrash fails on 32-bit machines when compiled without DEBUG_THREADS |
ASTERISK-22278: changemonitor returns WARNING[18526]: file.c:1229 ast_writefile permission denied when there is a permission |
ASTERISK-22279: when fxs(fxs is callee) is ring, dialing is set to 1,and so frametype = AST_FRAME_NULL in chan_dahdi.c/dahdi_read() , so will not invoke ast_dsp_busydetect() |
ASTERISK-22280: busydetect not work |
ASTERISK-22281: chan_pjsip tests: Complete off-nominal inbound call tests |
ASTERISK-22282: chan_pjsip tests: Uncomment and verify transport options in existing nominal/off-nominal tests |
ASTERISK-22283: chan_pjsip tests: Implement nominal outgoing call tests |
ASTERISK-22284: chan_pjsip tests: Implement off-nominal outgoing call tests |
ASTERISK-22285: chan_pjsip tests: Implement nominal Alice initiated two-party call tests |
ASTERISK-22291: ARI: /endpionts/{tech}/{id} channel list shouldn't have channel: prefixes |
ASTERISK-22292: res_stasis.c uses undefined function 'control_continue', maybe renamed to stasis_app_control_continue? |
ASTERISK-22293: tvfix(): Too large timestamp |
ASTERISK-22294: [patch] format_g729 - better handling of CNG |
ASTERISK-22296: ARI fails to find symbols for res_http_websockets |
ASTERISK-22297: Local channels, in a bridge, don't leave Stasis on hangup |
ASTERISK-22304: 'Bad Magic Number' with bridge Application after ~150 bridges |
ASTERISK-22306: res_pjsip endpoint config object's 'identify_by' option needs cleanup, removal or other modification |
ASTERISK-22308: Documentation - chan_dahdi, waitfordialtone is not boolean, it's time in milliseconds |
ASTERISK-22310: Improve error message 'Not a wav file '... in format_wav.c; current message is ambiguous |
ASTERISK-22311: [patch] 'identify' configObject doesn't have a synopsis |
ASTERISK-22313: handset get disconnected during a call |
ASTERISK-22314: Failure in canceling a call, sending OK to wrong port |
ASTERISK-22315: Asterisk 12 Test Suite Failures: Fix the Queue Tests |
ASTERISK-22316: Asterisk 12 Test Suite Failures: Fix the SIP one legged transfer test |
ASTERISK-22317: Asterisk 12 Test Suite Failure: Fix ConfBridge Nominal Test |
ASTERISK-22318: Asterisk 12 Test Suite Failure: Fix the Local Channel T.38 Query Option test |
ASTERISK-22319: Asterisk 12 Test Suite Failure: fix the asyncagi break test |
ASTERISK-22320: Asterisk 12 Test Suite Failures: pjsip one touch recording tests |
ASTERISK-22321: Asterisk 12 Test Suite Failures: inbound pjsip registration tests are failing |
ASTERISK-22322: Asterisk 12 Test Suite Failures: Fix the FastAGI Control Stream File test |
ASTERISK-22323: Asterisk 12 Test Suite Failures: Fix the "simple" bridge tests |
ASTERISK-22324: Asterisk 12 Test Suite Failures: Fix the feature transfer setup tests |
ASTERISK-22325: Asterisk 12 Test Suite Failures: Fix simple Bridge DTMF Feature Tests |
ASTERISK-22326: Asterisk 12 Test Suite Failures: Fix the dial L/S options test |
ASTERISK-22327: Asterisk 12 Test Suite Failures: Fix the Bridge Transfer Tests |
ASTERISK-22328: Asterisk 12 Test Suite Failures: Parking Tests |
ASTERISK-22344: Missing xml doc configOption 'type' for both 'system' and 'global' configObjects |
ASTERISK-22345: Realtime dialplan extension matching fails when extension includes "/" character |
ASTERISK-22346: core dump when processing call files; in ast_channel_tech_pvt ... at channel_internal_api.c |
ASTERISK-22347: [patch]res_xmpp timeout when google sends blank responses ' ' |
ASTERISK-22348: Action "sippeers" returns incorrect XML on Manager MXML interface |
ASTERISK-22349: Behavioral change in Asterisk 11: udptl now requires config file to function |
ASTERISK-22350: DUNDI - core dump on shutdown - segfault in sqlite3_reset from /usr/lib/libsqlite3.so.0 |
ASTERISK-22351: Segfault in LIBEDIT_INTERNAL after tgetstr(), when libncurses5-dev isn't installed |
ASTERISK-22352: [patch] IAX2 custom qualify timer is not taken into account |
ASTERISK-22353: Random Asterisk Segmentation Fault |
ASTERISK-22354: calleridname parameter of check_peer_ok is not used in 1.8 branch SVN |
ASTERISK-22356: Add equivalent AMI actions for the bridge CLI commands. |
ASTERISK-22358: Create documentation entries for parking |
ASTERISK-22359: Create documentation entries for res_pjsip's send_diversion and subminexpirey options |
ASTERISK-22360: Logging output from pjproject not sent through Asterisk logger |
ASTERISK-22365: [patch] chan_h323 can't be compiled |
ASTERISK-22366: Codec changes need to make the bridge channels compatible again. |
ASTERISK-22367: Rework CEL unit test verification step |
ASTERISK-22368: [patch] mixmonitor_free leaks filename |
ASTERISK-22371: features.conf config warnings/errors |
ASTERISK-22372: res_corosync: Compilation errors and functionality broken in Asterisk 12 |
ASTERISK-22374: Finish mapping the sip.conf parameters to res_sip.conf parameters |
ASTERISK-22376: [patch] memory leaks |
ASTERISK-22378: [patch] fix various memory leaks |
ASTERISK-22379: Strange dialplan lookups happening when dial()ing sip peers |
ASTERISK-22380: Inbound SIP call to a valid extension results in segfault in multicast_rtp_new at res_rtp_multicast.c |
ASTERISK-22382: modifying transport configuration results in a crash a few seconds after a 'core reload' (security_event_get_transport at res_pjsip/security_events.c) |
ASTERISK-22384: modifying transport configuration names results in non-functional transport after 'core reload' |
ASTERISK-22386: Outbound SIP registration, if the auth object's realm option is not set to the same value as the 401's realm, then we fail to create a new REGISTER with auth details |
ASTERISK-22388: Need debug indicating outbound registration attempt and success |
ASTERISK-22390: client_uri and server_uri config documentation lacks useful detail |
ASTERISK-22392: Fix the SIP refer_replaces_to_self test |
ASTERISK-22393: CEL: During BRIDGE_ENTER/BRIDGE_EXIT events, the Peer field is never populated, even if a channel is in the bridge |
ASTERISK-22394: crash when using localnet and external_signaling_address options, segfault in session_inv_on_tsx_state_changed at res_pjsip_session.c |
ASTERISK-22395: [patch] manager.c and res_agi.c leak results from ast_xmldoc_printable |
ASTERISK-22403: Provide support for detecting Fortress Style Payphone coins |
ASTERISK-22405: res_pjsip endpoint 'external_media_address' option needs doc clarification vs transport option of same name |
ASTERISK-22407: Error out at config parsing when two or more transports are set to bind on the same interface |
ASTERISK-22409: Local channels in a ConfBridge w/ jitterbuffer=yes leak ast_frame's after masquerade |
ASTERISK-22410: [patch] Change "Error isn't a PubSub error ..." error log to a debug log |
ASTERISK-22411: British English Sound Packs |
ASTERISK-22412: Memory corruption in cdr_custom.c. |
ASTERISK-22413: [patch] features.c TEST_FRAMEWORK leaks channel reference, preventing graceful shutdown |
ASTERISK-22414: [patch] voicemail and test_voicemail_api leaks |
ASTERISK-22415: asterisk-11.5.0 linohone 3.6.1 ice not work |
ASTERISK-22416: [patch] Segmentation fault (in process_applicationmap_line, at features.c) when using improper feature mapping syntax |
ASTERISK-22417: [patch]RTP ports left open after making calls using SIPTAPI |
ASTERISK-22424: bridge_native_rtp: Asterisk 12 attempts to remotely bridge on 200OK response to invite when the 200 lacks SDP |
ASTERISK-22426: features: Asterisk 12 fails to start without feature.conf present |
ASTERISK-22428: [patch] SIP unregister does not fully unregister when using Realtime sip peers and Expires not 0 on 200ok |
ASTERISK-22429: [patch] - chan_dahdi allows for updating both hw and sw gains, but dahdi show channel doesn't reflect changes |
ASTERISK-22430: AMI Originated calls are logged only if answered |
ASTERISK-22431: Reopened/Cloned - sendrecv in response to recvonly SDP, despite RFC 3264.6 allowed responses (sendonly and inactive only) |
ASTERISK-22432: Async AGI crashes Asterisk when issuing "set variable" command without args |
ASTERISK-22433: Asterisk not detecting DTMF for incoming calls over FXO Digium board. This was working earlier. |
ASTERISK-22435: [patch] jabber/xmpp MWI distributed pubsub issue where the mailbox and context get swapped at the remote end |
ASTERISK-22436: [patch] No BYE to masqueraded channel on INVITE with replaces |
ASTERISK-22437: Asterisk suddenly sends 300 Mbit/s of OPTIONS packets to its peers |
ASTERISK-22438: Asterisk 1.8.23.0 - Loop Registration |
ASTERISK-22439: Improve Asterisk Admin Guide output |
ASTERISK-22440: ARI - Update events.json to match latest Swagger specification |
ASTERISK-22441: WebSocket response when subprotocol is omitted violates spec |
ASTERISK-22445: res_pjsip_messaging: Message technology registers itself as SIP, preventing compatibility with chan_sip |
ASTERISK-22446: Build warnings when dev-mode disabled. |
ASTERISK-22450: No CLI response from database show |
ASTERISK-22451: ARI: Need the ability to subscribe to channels/endpoints |
ASTERISK-22453: [patch] chan_pjsip fails to unregister session supplement, can cause segfault |
ASTERISK-22454: Confbridge leaves channel and room open if hang up during name recording |
ASTERISK-22455: Asterisk 12 on Ubuntu Lucid deadlocks with DEBUG_THREADS+OPTIONAL_API enabled |
ASTERISK-22456: Logger.conf: Logging types ignored after specifying a verbose level |
ASTERISK-22457: Module load errors for test_ari_model.so |
ASTERISK-22458: XML config documentation improvements for res_pjsip_acl |
ASTERISK-22459: Compiling res_odbc against iODBC instead of unixodbc produces runtime errors |
ASTERISK-22467: [patch] memory leaks 1.8+ |
ASTERISK-22468: Asterisk crashes |
ASTERISK-22469: crash when res_jabber receives an XMPP IQ stanza with no 'from' |
ASTERISK-22471: Set default auth realm to challenge realm if auth object realm is empty |
ASTERISK-22474: res_pjsip / res_pjsip_session assertions and segfault |
ASTERISK-22477: array calleridname in the function check_user_full is too short |
ASTERISK-22478: [patch]Can't use pound(hash) symbol for custom DTMF menus in ConfBridge (processed as directive) |
ASTERISK-22480: Embedded pjproject: build.mak contains hardcoded full path to version.mak |
ASTERISK-22482: CDR Assertion failure when local channel leaves parking lot. |
ASTERISK-22483: AST_LIST_INSERT_TAIL doesn't set field.next on added entry |
ASTERISK-22485: ARI: Origination provides insufficient feedback |
ASTERISK-22486: ARI: TCP Reset after 204 response |
ASTERISK-22487: ARI: ARI Origination + SIP blind transfer == asterisk crash |
ASTERISK-22488: CDR performance bottleneck |
ASTERISK-22489: Document Confbridge record_file appending file name with time stamp. Behavior introduced in r381702 |
ASTERISK-22495: testsuite: ACL tests fail on Asterisk 11 |
ASTERISK-22498: [patch]Create functions to manipulate SIP headers when using PJSIP stack |
ASTERISK-22499: ARI documentation - point to HTTP server configuration sample and wiki docs where appropriate |
ASTERISK-22504: [patch] chan_iax2: wrong expiry time in astdb |
ASTERISK-22505: PJSIP cannot identify endpoints by dynamically-learned IP address |
ASTERISK-22506: Queue ringall strategy gets a CDR assertion failure for losing call. |
ASTERISK-22507: app_queue assertion failure on caller hangup |
ASTERISK-22513: astobj2.c INTERNAL_OBJ: bad magic number for 0x7f0b500174e8. Object is likely destroyed |
ASTERISK-22514: app_stasis missing silence generator |
ASTERISK-22517: ARI Client Libraries |
ASTERISK-22523: configured EuroISDN connected to PSTN not able to change TOn |
ASTERISK-22524: chan_ooh323 rely on system tcp timeout |
ASTERISK-22525: Realtime mysql database schema missing in contrib/realtime/mysql/ |
ASTERISK-22526: CDR fields specification does not correspond |
ASTERISK-22527: CDR issue on Attended transfer |
ASTERISK-22528: Change name of endpoint config option "external_media_address" to "media_address" |
ASTERISK-22530: Asterisk segfaults upon call transfer. |
ASTERISK-22531: Fix the sip_attended_transfer test |
ASTERISK-22532: Fix chan_pjsip two party alice initiated test failures |
ASTERISK-22536: MALLOC_DEBUG causes /tmp/refs to be written, even if REF_DEBUG is not defined |
ASTERISK-22537: Create Sorcery equivalent to the AST_CONFIG function |
ASTERISK-22538: Apparent loop in dlclose while loading dynamic modules during Asterisk startup |
ASTERISK-22539: Measure cellular signal strength of inbound calls |
ASTERISK-22540: WARNING[2324] asterisk.c: Fork failed: Cannot allocate memory in /var/log/asterisk/messages then segfault |
ASTERISK-22541: Very high CPU usage on Asterisk 11.5.1 |
ASTERISK-22542: Call parking test failure: test fails due to no translation path between IAX2 channel and Local channel |
ASTERISK-22543: Can only make one call (inbound or outbound) at a time. |
ASTERISK-22544: Italian prompt vm-options has advertisement in it |
ASTERISK-22546: Despite persistentmembers=yes, various queue member attributes set with AMI actions or QUEUE_MEMBER function are not stored in ASTDB |
ASTERISK-22548: [patch] Add SIP un-register tests |
ASTERISK-22551: Session timer : UAS (Asterisk) starts counting at Invite, UAC starts counting at 200 OK. |
ASTERISK-22552: Need log messages - res_pjsip_endpoint_identifier_user.so fails to find an endpoint match on inbound REGISTER |
ASTERISK-22554: Log message says we received a 408, when in reality we didn't receive anything. Clarify the log message. |
ASTERISK-22556: [patch] MailboxCount & ExtensionState commands - request multiple mailboxes/extensions |
ASTERISK-22557: [patch] Use waitpid instead of wait4 when we don't need to read from rusage |
ASTERISK-22558: Asterisk should not send re-invite and update during P-Asserted in call |
ASTERISK-22559: gcc 4.6 and higher supports weakref attribute but asterisk doesn't detect it. |
ASTERISK-22560: Memory leak in logger.c |
ASTERISK-22561: Open blockers for 1.8.24.0 |
ASTERISK-22562: Open blockers for 11.6.0 |
ASTERISK-22563: Realtime database connections dropping |
ASTERISK-22564: Local bridge in bridge_native_rtp causes one way audio |
ASTERISK-22565: [patch] res_rtp_asterisk leaks reference to rtcp_report in ast_rtcp_read |
ASTERISK-22566: [patch] app_cdr leaves application registered and res_parking leaks a ref to config |
ASTERISK-22567: [patch]MutlicastRTP does not set SSRC. SSRC is always set to 0 |
ASTERISK-22568: hard-wired sounds in app_meetme: enter and leave |
ASTERISK-22569: JABBER_RECEIVE function crash Asterisk |
ASTERISK-22570: [patch] xslt library cleanup |
ASTERISK-22571: [patch] Testsuite: pjsip diversion tests missing dependencies |
ASTERISK-22572: Asterisk 11.5.1- SPARC don't start due to many ast_symbols not found |
ASTERISK-22573: in Asterisk 12, cdr_mysql does not compile and makes the whole project fail |
ASTERISK-22574: [patch]Value of expires= is ignored in the Contact header |
ASTERISK-22575: Call from SIP phone to OOH323 - can't hide callerid |
ASTERISK-22576: [patch] Testsuite: SIP path test creates extra channels requiring asterisk to be killed |
ASTERISK-22577: Asterisk 11.5.1- SPARC don't start due to many ast_symbols not found |
ASTERISK-22578: Invalid manager logins aren't reported via security events: Invalid IE Specified ERROR |
ASTERISK-22579: peer is not matched to an IP address |
ASTERISK-22580: chan_pjsip: crash when attempting to add best codec to channel and no formats are found |
ASTERISK-22581: AMI: ConfbridgeList has race condition causing crashes |
ASTERISK-22582: [patch] chan_sip refactor - sip_route |
ASTERISK-22589: shared_lastcall is not reliable with realtime queues |
ASTERISK-22590: BufferOverflow in unpacksms16() when receiving 16 bit multipart SMS with app_sms |
ASTERISK-22591: [patch]Prevent Asterisk from writing received SMS content in log |
ASTERISK-22594: QUEUE membermacro args |
ASTERISK-22604: app_queue is dependent upon AMI subscribing to stasis. |
ASTERISK-22607: Dial application option 'r' does not use indications for the tonezone in use by channel, unless passing an argument |
ASTERISK-22608: [patch] Substitute Variables In Features Application Map Before Execution |
ASTERISK-22609: Implement AMI commands for PJSIP |
ASTERISK-22610: Implement CLI commands for PJSIP |
ASTERISK-22613: cdr_prop function is not working |
ASTERISK-22614: Asterisk 12 using 308.000 handles with 60 open calls |
ASTERISK-22615: sip_attended_transfer: crash on disposed of object in native RTP bridge |
ASTERISK-22616: timestring set by TESTTIME is ignored by IFTIME as well as GotoIfTime |
ASTERISK-22617: IFTIME doesn't support Timezoone-Parameter and still uses | |
ASTERISK-22619: [patch] Session-ID header support for chan_sip |
ASTERISK-22620: GotoIf in 'h' extension works incorrectly |
ASTERISK-22621: chan_sip can send two BYEs for a single call |
ASTERISK-22622: Disabled CEL generates ERROR message on call hangup. |
ASTERISK-22623: ARI: Recording using name of already present stored recording does not indicate failure |
ASTERISK-22624: ARI: Adding a channel to a bridge while a live recording is active blocks |
ASTERISK-22625: Core dump on ARI dial - ast_copy_string at .../include/asterisk/strings.h |
ASTERISK-22626: ARI: Starting a recording with invalid format does not report error or failure |
ASTERISK-22627: ARI: Calling record on a non-existent bridge causes response validation failure |
ASTERISK-22628: 4 way multi-party hanging up down to two participants causes FRACKs |
ASTERISK-22629: Bridge hangs around when DTMF feature parkcall not available |
ASTERISK-22630: When a parked call times out the original parker cannot initiate transfers after pickup. |
ASTERISK-22631: Reload does not rebuild parkpos extensions |
ASTERISK-22632: option allowexternaldomains behavior changed between release versions |
ASTERISK-22633: Patch to add a new channel variable SIPURIOPTION to save the URI options for call transfer target |
ASTERISK-22634: ARI: can't delete a bridge with a channel in it. |
ASTERISK-22635: ARI: listing bridges shows bridges that are not part of an ARI application. |
ASTERISK-22636: ARI: swagger-ui DELETE does not allow entering channelid. |
ASTERISK-22637: ARI channel mute causes core |
ASTERISK-22644: Crash with app queue and DND set on SIP agent phone |
ASTERISK-22645: Broad media offers from Jitsi client results in a crash in ast_copy_pj_str at res_pjsip.c |
ASTERISK-22658: PJSIP: If a transport is set on an endpoint, Asterisk will not reuse established connections for that endpoint |
ASTERISK-22659: Make a new core and extra sounds release |
ASTERISK-22661: Unable to exit ChanSpy if spied channel does not have a call in progress |
ASTERISK-22662: Documentation fix? - queues.conf says persistentmembers defaults to yes, it appears to lie |
ASTERISK-22666: Outbound proxy ignored after initial invite |
ASTERISK-22667: crash: directmedia with both phones placing each other on hold |
ASTERISK-22668: Crash: chan_pjsip extension fallthrough |
ASTERISK-22669: AMI/CLI Agent Logoff with soft logs out agent on subsequent logins |
ASTERISK-22670: Asterisk crashes when processing ISDN AoC Events |
ASTERISK-22671: AUDIOHOOK_INHERIT Fails on Outbound Attended Transfer |
ASTERISK-22672: Crash when dialing back to same device from dialplan when endpoint have outbound_proxy set. |
ASTERISK-22673: OPTIONS packets doesn't comply with endpoint outbound_proxy setting |
ASTERISK-22674: [patch]add a channel variable for recording status in mixmonitor application |
ASTERISK-22675: Asterisk refuses correct RTP/AVP with optional encryption |
ASTERISK-22676: Native RTP (p2p) bridge is not torn down during transition to soft mix |
ASTERISK-22677: Playbacks on bridge via ARI are not queued |
ASTERISK-22678: crash: play non-existent sound to bridge |
ASTERISK-22679: crash: play unspecified sound to bridge |
ASTERISK-22680: crash: playback control with operation omitted in POST |
ASTERISK-22681: RES XMPP |
ASTERISK-22682: No dtmf cognition of features.conf after Ami Action Bridge |
ASTERISK-22683: SipUpdate on Local Channel will crash asterisk |
ASTERISK-22684: extenpatternmatchnew doesn't match extensions that contain '-' |
ASTERISK-22685: Unable to POST content-type: application/json to ARI |
ASTERISK-22686: Asterisk does not sends RTP when transfer is done in telco side |
ASTERISK-22687: Core Dump On DB Connection Failing with MariaDB |
ASTERISK-22688: chan_sip: send_provisional_keepalive_full called on destroyed SIP pvt resulting in crash |
ASTERISK-22695: ARI: Add the ability to monitor a channel |
ASTERISK-22697: ARI: Add the ability to raise an arbitrary User Event from the Asterisk or Applications resource |
ASTERISK-22699: ARI: Add the ability to issue log messages to the Asterisk logging facilities |
ASTERISK-22701: ARI: Add the ability to indicate things to channels |
ASTERISK-22703: crash: hanging up channels in a softmix bridge |
ASTERISK-22704: ARI: allowMultiple parameters should generate appropriate docs |
ASTERISK-22705: ARI: wiki docs for the models are screwy |
ASTERISK-22706: No dtmf cognition of features.conf after Ami Action Bridge |
ASTERISK-22707: SipUpdate on Local Channel will crash asterisk |
ASTERISK-22708: res_odbc.conf negative_connection_cache option not respected, failover between DSNs doesn't work |
ASTERISK-22709: crash: atxfer threeway call results in crash while creating channel snapshot |
ASTERISK-22710: ARI: Media operations (playback, recording, etc) need to push events back over the websocket |
ASTERISK-22717: SRTP audio stream rejected, 'Could not set SRTP policies' |
ASTERISK-22718: Devstate caching mangled after local channel optimization |
ASTERISK-22719: ARI: Call forwarding is not reflected or followed on origination |
ASTERISK-22720: devstate incorrectly cached on transfer of a DAHDI to SIP call. |
ASTERISK-22721: crash: possible ref counting problem in Stasis or in one of the consumers |
ASTERISK-22722: ARI: GET/DELETE /playback/{invalid-id} has response validation failure. |
ASTERISK-22727: ARI - DELETE global variables |
ASTERISK-22728: [patch] Improve Understanding Of 'Forcerport' When Running "sip show peers" |
ASTERISK-22729: [patch] Remove Port Restriction When Checking For NAT |
ASTERISK-22731: Crash on incoming chan_pjsip call where dialplan hangs up before ACK is received for INVITE |
ASTERISK-22732: Deadlock potential in res_fax and CCSS with local channels. |
ASTERISK-22737: Rename ARI Playback to Playbacks |
ASTERISK-22738: "Security denial" error in calls from H323 trunk (ooh323.c) |
ASTERISK-22739: Calls are bill before the connect |
ASTERISK-22740: [patch] - Confbridge fails to destroy conference on hangup leading to Asterisk segfault |
ASTERISK-22741: In chan_iax2 multiple addresses can no longer be bound to |
ASTERISK-22743: ARI: Allow POST parameters to be submitted in the request body |
ASTERISK-22744: ARI: Hidden channels show up in bridge snaphsots and bridge events |
ASTERISK-22745: chan_sip call setup very slow or fails when STUN server not available |
ASTERISK-22746: [patch]Crash in chan_dahdi during caller id read |
ASTERISK-22748: SRTP Crypto Offer With Lifetime Not Accepted |
ASTERISK-22749: Deadlock during 4-way conference creation |
ASTERISK-22750: SIP TLS calls stop working after a period of no SIP TLS calls to a destination |
ASTERISK-22755: Memory leak in voicemail module with mailbox_full var |
ASTERISK-22756: [patch]Correct cause code for chan_dahdi request when trunk congested |
ASTERISK-22757: segfault in res_clialiases.so on reload when mapping "module reload" command |
ASTERISK-22758: [patch]Support for A-law and u-law WAV file format |
ASTERISK-22759: ARI: DELETE on /bridges/{id} fails to remove bridge even with 204 success |
ASTERISK-22760: ConfBridge: CONFBRIDGE function does not allow for dynamic creation of menu items |
ASTERISK-22763: Segfault in __ao2_find () |
ASTERISK-22765: [patch]South Africa end of call detection |
ASTERISK-22768: ARI: Originating multiple channels using POST /channels in succession causes orphaned channels; other problems |
ASTERISK-22777: pjsip messaging: Investigate in dialog message request test failures; determine correct behavior of in call messaging |
ASTERISK-22780: ARI: Implement channel spying |
ASTERISK-22781: ARI: Implement channel monitoring |
ASTERISK-22784: ARI: Kill /dial |
ASTERISK-22786: Asterisk crash with "Bus Error" when users hangup after leaving voicemail (ODBC storage) |
ASTERISK-22787: Call file RetryTime not respected after several retries |
ASTERISK-22788: [patch] main/translate.c: access to variable f after free in ast_translate() |
ASTERISK-22789: chan_sip SDP incorrect negotiation for RTP payload types causes DTMF recognization failure and appears to violate RFC |
ASTERISK-22790: check_modem_rate() may return incorrect rate for V.27 |
ASTERISK-22791: asterisk sends Re-INVITE after receiving a BYE |
ASTERISK-22793: Base64 encoding not honoring \r\n |
ASTERISK-22796: Documentation: add see-also links in AMI event documentation for event pairs |
ASTERISK-22797: [patch]Default values for many iax.conf options , including auth= and trunktimestamps= are not indicated in /configs/iax.conf.sample |
ASTERISK-22799: CEL: Invalid cel.conf will fail to create Stasis topic, allowing for a crash if CELGenUserEvent attempts to publish to it |
ASTERISK-22800: Dynamic parking pickup doesn't use PARKINGLOT |
ASTERISK-22801: Memory Corruption during /pjsip/basic_calls/incoming/nominal/unauthed/ident_by_host testsuite test |
ASTERISK-22802: MeetMe: crash when ast_check_hangup is called on a NULL conference channel |
ASTERISK-22803: ari: GET /ari/endpoints/{invalid-tech} should return a 404 |
ASTERISK-22804: Redirecting two bridged lines via AMI causes race condition, preventing the second redirect from working properly. |
ASTERISK-22805: res_rtp_asterisk: Crash when calling BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP |
ASTERISK-22814: GROUP broken for attended transfer (Both SIP transfer and features.conf) |
ASTERISK-22815: Alembic does not create realtime dialplan tables |
ASTERISK-22817: ooh323 reload command doesn't send request for registreation to gk |
ASTERISK-22818: res_pjsip distributor: invalid access to PJSIP mutex results in crash |
ASTERISK-22820: [patch] Plaintext auth is still supported in IAX2 |
ASTERISK-22821: Asterisk 12-beta @r402448 pjsip sigsegv receiving SIP MESSAGE when checking Contact header |
ASTERISK-22822: Recieving SMS not working |
ASTERISK-22825: Dialplan Function for Checking Parking Lot Slot |
ASTERISK-22826: cdr_adaptive_odbc SQL execute error with PostgreSQL |
ASTERISK-22827: documentation: A number of ManagerEvent docs are inconsistent in 11 |
ASTERISK-22829: AMI Atxfer not working with alphanumeric extensions |
ASTERISK-22830: Change log output for audio files with bad permissions |
ASTERISK-22831: Commas cannot be used as part of any string being passed to ODBC as a SQL parameter, due to lack of string field encapsulation |
ASTERISK-22832: Support AES-GCM mode in SRTP |
ASTERISK-22833: channel: Channel reference leak |
ASTERISK-22834: Parking by blind transfer when lot full orphans channels |
ASTERISK-22835: pbx_realtime: deadlock with channel in autoservice while calling realtime switch |
ASTERISK-22838: ARI: Implement device state API |
ASTERISK-22841: Improve documentation of t1min by clarifying its behavior as a global T1 minimum setting |
ASTERISK-22842: V29 is incompatible with minrate setting 2400 |
ASTERISK-22843: testsuite: Sporadic failure of bridge_transfer_callee test on Asterisk 12+ |
ASTERISK-22845: An extension in use receive more than one call with leastrecent strategy |
ASTERISK-22846: testsuite: masquerade super test fails on all branches (still) |
ASTERISK-22851: Asterisk/SIP+RTP stops responding when compiled with DEBUG_THREADS |
ASTERISK-22852: Error decoding H245 message - ooh323c |
ASTERISK-22853: SIP call hangup randomly during conversation due to MixMonitor |
ASTERISK-22854: [patch] - Deadlock between cel_pgsql unload and core_event_dispatcher taskprocessor thread |
ASTERISK-22856: [patch]SayUnixTime in polish reads minutes instead of seconds |
ASTERISK-22857: Deadlock: Locked Here: chan_iax2.c line 9756 (socket_read), when compiled with DEBUG_THREADS |
ASTERISK-22858: Crash in chan_pjsip/PJSIP stack when unsupported codec is specified in allow |
ASTERISK-22859: Deadlock random |
ASTERISK-22860: overlap dial doesn't work, TE420-panasonic kx-td500 |
ASTERISK-22861: [patch]Specifying a null time as parameter to GotoIfTime or ExecIfTime causes segmentation fault |
ASTERISK-22862: app_meetme wait_for_leader - charging for the actual duration of the conference, not for all call to meetme |
ASTERISK-22864: Queue agent order when changing to linear strategy not correct on reload |
ASTERISK-22865: externhost is not work |
ASTERISK-22866: Asterisk 12 (branches/12@402864) doesn't correctly set the fromuser parameter |
ASTERISK-22868: chan_pjsip: 'setvar' should be supported on endpoints |
ASTERISK-22870: dialplan entries pointing to SIP peers not defined in sip.conf just hangs the call |
ASTERISK-22871: cel_pgsql module not loading after "reload" or "reload cel_pgsql.so" command |
ASTERISK-22872: ARI: Allow specifying channel variables during a POST /channels |
ASTERISK-22874: CDR Lines Missing |
ASTERISK-22875: CLONE - Segfault in __ao2_find () |
ASTERISK-22882: PJSIP + Blink + ARI mixing bridges => unexpected calls coming into blink |
ASTERISK-22884: hangup_handler end with h extension: tests currently fail in Asterisk 12 + |
ASTERISK-22885: bridge/transfer_failure test failure |
ASTERISK-22886: CDRs: Applications that manipulate CDRs are out of step with engine, creating unpredictable results |
ASTERISK-22887: pjsip tests: interactions with chan_sip cause test failures |
ASTERISK-22888: RES Corosync |
ASTERISK-22889: Segmentation fault when RTP going via ICE |
ASTERISK-22890: pjsip inbound registration nominal test: Crash during memcpy in pjsip_print_msg |
ASTERISK-22891: One way audio if dialplan_exec menu option runs Dial application and certain codecs are used |
ASTERISK-22897: WebSocket connection from JsSIP or SIPML5 generate a segmentation fault(core dumped) |
ASTERISK-22899: Manager UserEvent including action on output |
ASTERISK-22900: chan_sip miss parsed realm, if contain @ character |
ASTERISK-22901: Originate command doesn't work through OOH323 channels |
ASTERISK-22902: Crash when setting RTCP property on RTP instance |
ASTERISK-22903: SIP Hangs, Asterisk becomes unresponsive when compiled with DEBUG_THREADS |
ASTERISK-22904: bridges: lock the bridge when creating bridge snapshots |
ASTERISK-22905: Prevent Asterisk functions that are 'dangerous' from being executed from external interfaces |
ASTERISK-22909: Less Cryptic security_events.c output |
ASTERISK-22910: [patch] - REPLACE() calls strcpy on overlapping memory when <replace-char> is empty |
ASTERISK-22911: [patch]Asterisk fails to resume WebRTC call from hold |
ASTERISK-22912: res_corosync doesn't build in Asterisk 12 beta2 |
ASTERISK-22918: dahdi show channels slices PRI channel dnid on output |
ASTERISK-22919: core show channeltypes slicing |
ASTERISK-22920: Crash while Forwarding from TLS extension with CHANNEL args secure_bridge_media and secure_bridge_signaling |
ASTERISK-22922: Asterisk 12 (branches/12@403134) doesn't load some endpoints |
ASTERISK-22923: module reload res_pjsip.so core dumps |
ASTERISK-22924: PJSIP MESSAGE support does not present the contact information on outbound messages |
ASTERISK-22925: Asterisk Crash |
ASTERISK-22928: Kick all users Confbrige |
ASTERISK-22929: queue incomming connect, Attended call transfer doesn't get back (on busy or noanswer) |
ASTERISK-22930: IAX constantly changing default port 4569 |
ASTERISK-22931: Impossible to execute Asterisk because if illegal instruction |
ASTERISK-22932: [patch] - SIP Channel fails to parse refer_to_domain |
ASTERISK-22933: Preventig RT packets to public IP's: res_rtp_asterisk.c:3574 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address |
ASTERISK-22934: Asterisk stops processing sip messages |
ASTERISK-22935: [patch] Infinity retransmission SIP packet loop when no response from far end |
ASTERISK-22936: Deadlock during masquerade when a PJSIP channel attended transfers a 3+ party bridge to dialplan |
ASTERISK-22937: Asterisk crashes with pj_NO_MEMORY_EXCEPTION exception |
ASTERISK-22938: Asterisk Crashes with assert fail for 'ype <= PJ_ICE_CAND_TYPE_RELAYED' |
ASTERISK-22939: [patch] Missing/wrong local party in dialog-info NOTIFY body |
ASTERISK-22940: AGI - 'h' Extension Runs When FAX Is Detected, Not At Hangup |
ASTERISK-22942: [patch] - Asterisk crashed after Set(FAXOPT(faxdetect)=t38) |
ASTERISK-22944: Reload Memory Leak. |
ASTERISK-22945: [patch] Memory leaks in chan_sip.c with realtime peers |
ASTERISK-22946: Local From tag regression with sipgate.de |
ASTERISK-22951: Wrong RTP timestamps (resulting in audio issues) when transcoding to SILK from G711 |
ASTERISK-22952: res_pjsip_pubsub: crash when subscription_destructor is terminated from a non-PJSIP thread |
ASTERISK-22954: [patch] Incorrect treatment of amaflags, accountcode and userfield since ASTERISK-16990 |
ASTERISK-22956: core prompts for Russian set require a few new files |
ASTERISK-22958: app_queue members state interface has issues when using a hint through an included context |
ASTERISK-22959: ARI /asterisk/getGlobalVar doesn't execute functions |
ASTERISK-22960: [patch] Segfaults in res_musiconhold.c:moh_release |
ASTERISK-22961: [patch] DTLS-SRTP not working with SHA-256 |
ASTERISK-22962: performance spike on Local channels originated using ARI |
ASTERISK-22970: [patch]Documentation fix for QUOTE() |
ASTERISK-22972: [patch] CLI "manager show commands" Truncates Columns |
ASTERISK-22976: app_queue function queue_show() and find_queue_by_name_rt() cause deadlock |
ASTERISK-22977: chan_sip+CEL: missing ANSWER and PICKUP event for INVITE/w/replaces pickup |
ASTERISK-22980: [patch]Allow building cdr_radius and cel_radius against libfreeradius-client |
ASTERISK-22981: Asterisk crashes while look up AMI action in registered action list |
ASTERISK-22982: CEL/cel_custom: Using non-standard func_channel CHANNEL(..) variables causes segfaults |
ASTERISK-22983: DAHDI channel gets stuck in 'Pre-rin' state |
ASTERISK-22984: ari: Transfer messages not being sent out ARI WebSocket |
ASTERISK-22985: System() doesn't handle echo -e like it should |
ASTERISK-22987: No handler for event dialog BLF |
ASTERISK-22988: [patch]T38 , SIP 488 after Rejecting image media offer due to invalid or unsupported syntax |
ASTERISK-22991: pjsip/ami tests: tests bouncing |
ASTERISK-22992: [patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel |
ASTERISK-22993: pjsip/basic_calls/two_parties/nominal tests bouncing |
ASTERISK-22994: pjsip/incoming_calls_without_auth: test bounces |
ASTERISK-22995: tests/rest_api: unhappy bridge test bounces |
ASTERISK-22996: tests/rest_api: applications/subscribe-bridge fails |
ASTERISK-22997: tests/rest_api: channels/originate fails |